[OpenSIPS-Users] Incorrect tls port

Wed Jan 5 14:43:18 UTC 2022

Bogdan, thanks a lot for your replies!

Best Regards,
Sergey Pysanko.

On Wed, Jan 5, 2022, 16:37 Bogdan-Andrei Iancu <bogdan at opensips.org> wrote:

> I mean, as per SIP, the UAS device must mirror, without any changes, the
> received RR into the 200 OK replies. And here even if Asterisk receives the
> RR hdr with the 5061 port, it sends back a 200 OK with a 48470 port in RR
> :-/
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
> OpenSIPS eBootcamp 2021
>   https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>
> On 1/5/22 4:32 PM, Sergey Pisanko wrote:
>
> Bogdan.
>
> Is it refers to the specific Asterisk behaivior scheme below? Asterisk's
> ACK of leg 2 and 200 OK of leg1 must be addressed to Opensips port 5061?
>
> Best Regards,
> Sergey Pysanko.
>
> On Wed, Jan 5, 2022, 15:54 Bogdan-Andrei Iancu <bogdan at opensips.org>
> wrote:
>
>> Hi Sergey,
>>
>> If Asterisk is the one changing (from 5061 to 48470) the port in the
>> RR/Route header, that's illegal to do.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   https://www.opensips-solutions.com
>> OpenSIPS eBootcamp 2021
>>   https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>>
>> On 1/5/22 10:48 AM, Sergey Pisanko wrote:
>>
>> Hi, Bogdan.
>>
>> Yes, you are right. That's full call's scheme.
>>
>> Opensips:48470                                 Asterisk (5062)
>> 1 leg ------------------INVITE (RR:5061)------------>
>> <-----------------INVITE--------------------------------- 2 leg
>> 2 leg --------------OK (RR:5061)-------------------->
>> <--------------------ACK (Route:48470)------------ 2 leg
>> < -------------------OK (RR: 48470) ----------------- 1 leg
>> 1 leg. ACK From UA1 to Asterisk through Opensips (Route:48470) sent, but
>> dropped.
>>
>>
>> Best Regards,
>> Sergey Pysanko.
>>
>>
>>
>> [image: Mailtrack]
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>> 10:45:28 AM
>>
>> вт, 4 янв. 2022 г. в 20:44, Bogdan-Andrei Iancu <bogdan at opensips.org>:
>>
>>> Sergey,
>>>
>>> I see OpenSIPS sents to Asterisk in INVITE:
>>>
>>> Record-Route:
>>> <sip:Opensips_IP:5061;transport=tls;lr;ftag=d8e0d49a268d4b51aa85b8f79d2dc062>
>>>
>>> but in the 200 reply from Asterisk back to OpenSIPS I see:
>>>
>>> Record-Route:
>>> <sip:Opensips_IP:48470;transport=TLS;lr;ftag=d8e0d49a268d4b51aa85b8f79d2dc062>
>>>
>>> Is asterisk the once changing the port there ???
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>>
>>> OpenSIPS Founder and Developer
>>>   https://www.opensips-solutions.com
>>> OpenSIPS eBootcamp 2021
>>>   https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>>>
>>> On 1/4/22 3:11 PM, Sergey Pisanko wrote:
>>>
>>> Hi, Bogdan.
>>>
>>> Here is my simple scenario description:
>>>
>>> UA1----Opensips----Asterisk ---- Opensips ----UA2
>>>
>>> Transport protocol doesn't change during this chain and it's tls, if I
>>> understand you right.
>>>
>>> I attached SIP capture of the call. As you can see, there is the
>>> dynamic tcp port in the RR hrd of last reply to client from which Opensips
>>> connected to the Asterisk. Instead of one, to which UA1 connected to
>>> Opensips (5061). As a result, there is a media session between UAs, but
>>> only for 30 sec, during of which the UA1 tried to send ACK to the Opensips,
>>> but unsuccessfully for quite clear reason. Is there the resolution how to
>>> realize this scenario without rewriting RR?
>>>
>>> Best Regards,
>>> Sergey Pysanko.
>>>
>>>
>>>
>>>
>>>
>>>
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>>> notified by
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>>> 01:46:49 PM
>>>
>>> вт, 4 янв. 2022 г. в 11:47, Bogdan-Andrei Iancu <bogdan at opensips.org>:
>>>
>>>> Hi Sergey,
>>>>
>>>> Manually altering the RR hdr is a receipt for disaster :). Somehow I
>>>> suspect you do not do double RR (as the protocol changes for the call).
>>>> This double RR is automatically done (by default) when doing
>>>> `record_route()`. Do you get 2 RR hdrs when routing the initial INVITE ?
>>>>
>>>> Regards,
>>>>
>>>> Bogdan-Andrei Iancu
>>>>
>>>> OpenSIPS Founder and Developer
>>>>   https://www.opensips-solutions.com
>>>> OpenSIPS eBootcamp 2021
>>>>   https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>>>>
>>>> On 1/4/22 11:27 AM, Sergey Pisanko wrote:
>>>>
>>>> Hello, Bogdan, .
>>>>
>>>> Thank you for your answer. I've solved my issue recently just rewriting
>>>> Record - Route header with appropriate port within "onreply route block" by
>>>> subst function.
>>>>
>>>> Best Regards,
>>>> Sergey Pysanko.
>>>>
>>>>
>>>>
>>>> [image: Mailtrack]
>>>> <https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&> Sender
>>>> notified by
>>>> Mailtrack
>>>> <https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&> 01/04/22,
>>>> 11:27:07 AM
>>>>
>>>> пн, 3 янв. 2022 г. в 17:59, Bogdan-Andrei Iancu <bogdan at opensips.org>:
>>>>
>>>>> Hello Sergey,
>>>>>
>>>>> Could you provide a SIP capture (and calling scenario) to underline
>>>>> the issue you have ?
>>>>>
>>>>> Best regards,
>>>>>
>>>>> Bogdan-Andrei Iancu
>>>>>
>>>>> OpenSIPS Founder and Developer
>>>>>   https://www.opensips-solutions.com
>>>>> OpenSIPS eBootcamp 2021
>>>>>   https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>>>>>
>>>>> On 12/30/21 2:50 PM, Sergey Pisanko wrote:
>>>>>
>>>>> Hello!
>>>>>
>>>>> I try to realize the next scenario with UAs, Opensips-2.4 and Asterisk.
>>>>> UAs are registered onto Asterisk through Opensips and also - on
>>>>> Opensips if the 200 OK is came back from Asterisk.
>>>>> Calls between UAs are relayed to Asterisk by Opensips.
>>>>> This scenario works fine with udp. But it needs to do with tls. And
>>>>> here I have the problem. What happens.
>>>>> Unlike udp, tcp cannot listen its port and create clients connection
>>>>> at the same time. Opensips listens tls port for clients connection
>>>>> whereas it creates dynamic tcp port to connect to Asterisk. As a
>>>>> result, I see that port in Record-Route header in 200 OK addressed to
>>>>> caller.
>>>>> Thus, callers ACK comes to that dynamic port instead of Opensips
>>>>> listened port and Opensips dropped it.
>>>>> And question is how to force Opensips to put right port for caller?
>>>>>
>>>>> Regards,
>>>>> Serhii Pysanko.
>>>>>
>>>>>
>>>>>
>>>>> [image: Mailtrack]
>>>>> <https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&> Sender
>>>>> notified by
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>>>>> 02:49:47 PM
>>>>>
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